Messages - bulek

I remember that I had to fix few things and got telephony in some kind of "working state". I think that some of basic Asterisk debugging skills should be used... First of all, enter asterisk shell with "asterisk -vvvvvvvvvvvvvvvrgc" and from there you can check "sip show registry" state of your SIP trunks, then "sip show peers" your extensions, etc...

Then also in shell you can get debug messages when you try to make a call - then try to locate warnings, errors. You can even get more messages by "tail -f /var/log/asterisk/messages"

I've found my notes and instructions what I needed to fix my system - hope they will help, although they should get into updates soon :

; ###########################################################################; PLUTOs "trusted" context; ###########################################################################[trusted]include => from-internal[19.2.2012 3:12:25] tiniahouse: comments :[19.2.2012 3:14:05] tiniahouse: I have SIP trunk to my provider on dedicated network card... For some reason, all incoming calls (even from trunk) end in from-sip-external context that is basically meant for anonymous incoming calls... so it's pretty restricted and all I got was "out of service" annoucement...[19.2.2012 3:15:47] tiniahouse: So I created my own context (first two lines) and also made this context default. That lowers security level, but we already have passwords being the same as extension number... Now, when call comes in, if it's for DID, then it goes to coresponding incoming route - that's first step I made[19.2.2012 3:17:15] tiniahouse: Now, my telephony is organized in this manner : I route all incoming calls to one dummy Dianemo user (housephone)... Why ? Cause I don't need to have more users and then I have to switch statuses for all of them... So we have housephone and calls are routed according to its state...[19.2.2012 3:19:18] tiniahouse: so at the end of incoming route, there is a dial for Local channel 307@.... and 307 is virtual extension for user housephone... That was not working cause 307 and all other user extensions were not visible in dialplan... For some reason file with them is not included in any other config files.... So that explains 3rd line[19.2.2012 3:21:18] tiniahouse: Then I got to proper housephone user, but calling more users at once (according to routing setup in admin page) was not working, cause calls are made as 202@trusted - trusted context was unknown in my system (it's only synonim for context from-internal - so that explains last lines[19.2.2012 3:22:33] tiniahouse: I also had to change /etc/asterisk/sip_general_custom.conf ,where I changed from-sip-external to my new context incoming[19.2.2012 3:22:38] tiniahouse: -sip-calls....[19.2.2012 3:23:34] tiniahouse: Also, there are files missing in /usr/share/asterisk/agi-bin - those with pluto-... in front of them (I compared to my 7.10 system)...[19.2.2012 3:23:57] tiniahouse: Now also incoming calls are working ok in my house... Did you follow me ?

I have (among others) a 7970, this is the sscp.conf. The 7970 has 2 lines. One especially for a door bell.

....

Thanks for response and info. How is your doorbell connected to asterisk - is it just ordinary extension ? Is this setup different from setup where doorbell call comes in and goes to other phones by nornal call routing ?

I'm using squeezeslave that is amplified in MArantz receiver on multiroom channel and then connected to Speaker selector ESS. I'm currently using it in this way :- I've put squeezeslave in certain room. When I start playing, MArantz is also properly switched on and set to proper input. But I have to manually enable speakers that are active in speaker selector...

I wonder if I could set up Squeezeslave in such manner, that it would be visible as player in 4 entertainment areas and when I start playing in certain zone, everything should be setup automatically (including ESS) ?

I've 3 7970 phones and they worked quite ok under 7.10. I'd like them to work under newer versions of LMCE - they also support a lot of various features that can be put in config file. But when I take a look at this config files, they all seem a bit cryptic for me.

Does anyone know of any document that would (shortly) explain features & settings of sccp channel under asterisk ?

I'm setting up Dianemo NC with mythtv 0.24. I have IPTV streams available that in MPEG2/TS format some time ago. Under 7.10, mythtv was unable to receive UDP streams, so I had to put vlc in between to change format of the streams into rtsp container. That worked, I was able to see iptv channels on mythtv and also on any PC with vlc client.

Recently, my iptv provider switched to MP4 streams (it seems that still in TS container) and mythtv was not working anymore under 7.10. Now I'm on Dianemo S10.10 and mythtv 0.24 and I wonder if there is any better chance to receive those streams also in mythtv...

Hmm, this is most strange.The server had malfunctioned for some days when I wrote my question. Being away for some days, and today I picked the server out of the closet to do some investigations as suggested above. Power on - and everything works....

So I seem to have a self healing server, which in one way is very nice, but I hate to have these situations when I cannot find the cause of the fault. Most likely it'll come back again.Thanks for you good suggestions, I'll save then until it's needed again. ;-)

Hi,

I could post similar thread but with 7.10 version.... Sometimes it just goes mad and settles down after few days....I've seen some strange warnings about sql transactions taking too much time, but I guess my database is also messy after all these years of daily usage.

I've regenerated media database and things seem to be a bit better than before (my living room MD crashed DCERouter when I went into Audio or Video entries on UI2, so I guess this was in relation with media database being messed) ...

I have followed procedure on 7.10 and got only partial success. I've managed to shutdown core services, dropped database, imported fresh one. Enabled daemon, but the problem is that after 3 days, if I press Filename option in Video or Audio menus I get empty screen - seems like no media file is present...

Through web-admin I can find files that have Location filled - for instance "/home/public/data/videos/Media [41]/Funny/name_movie.mp4" but they don't appear on Orbiters...